Heretofore, before a digital audio signal is supplied to a digital-to-analog converter, oversampling processing is performed to severalfold convert a sampling frequency from the original value. Therefore, in a digital audio signal outputted from the digital-to-analog converter, the phase characteristic of an analog anti-alias filter is kept at the upper area of an audio-frequency, and the influence of digital image noise accompanied with the sampling is removed.
In the above oversampling processing, generally, a digital filter by linear interpolation system of first degree is applied. Such digital filter generally generates linear interpolation data by obtaining the mean value of plural existent data when sampling rate has changed or data has defected.
Although the data quantity of the digital audio signal after oversampling processing becomes accurate severalfold in the time axis direction by linear interpolation of first degree, however, the frequency band of the digital audio signal after oversampling processing is almost the same as before conversion; the sound quality itself is not improved. Furthermore, since all of the interpolated data were not generated based on the waveform of the analog audio signal before A/D conversion, the reproducibility of waveform is scarcely improved.
On the other hand, when digital audio signals having a different sampling frequency are dubbed, the frequency is converted with a sampling rate converter. In such case, however, to improve the sound quality and the reproducibility of waveform have been difficult because only linear interpolation of data by a linear primary digital filter cannot be performed. It is similar to the case where the data sample of the digital audio signal has defaulted.